Multi-mode coding method is a reliable DC-suppression method. There are two ways to improve the DC-suppression performance. One is improving scrambler's performance, and the other is improving selection criteria. The latter uses the MRDS(minimum running digital sum) criterion. It is easy to calculate, but its performance goes down when the length of codeword is getting longer. The MSW(mean squared weight) criterion that is known as the best so far regardless of the length of codeword has the high complexity. In this paper, we present the new selection criteria, MPRDS(minimum peak RDS) and A BSRDS(absolute RDS). Their performance are close to the MSW, implementation is simple. And also we present the SC(sign change) that has a subsidiary role with the original selection criteria and improve the capacity.

Turbo codes are selected as FEC(Forward error correction) codes with convolution code in 3GFP(3rd generation partnership project) and 3GPP2 standard of IMT2000. Especially, l/3 turbo code with K=4 is employed for 3GPP standard. In this paper, we proposed a hardware structure of a turbo decoder and denveloped the decoder for 3GPP standard turbo code. For its efficient operation, we design a SOVA decoder by employing a register exchange decoding block and new path metric normalization block as a SISO constituent decoder. In addition, we estimate its performance under MATLAB 6.0 and designed the turbo decoder including control block, input control buffer, SOVA constituent decoder with VHDL. Finally, we synthesized the developed turbo decoder under Synopsys FPGA Express and verified it with ALTERA EPF200SRC240-3 FPGA device.

In this paper we proposed system parameter values of ultra-wideband Impulse Radio systems for the frequency band(3.1~10.6GHz), which is allocated by Federal Communications Commission(FCC). We also analyzed performance of the proposed system in the multiple access interference environment. According to result, application of possible pulse duration() is very limited by 0.04~0.0326 ns in permission frequency range that establish in FCC. In the case of the same pulse signal power, we could know that system performance was changed by pulse repetition number( ) regardless of pulse duration. Thus, We could know that we have to need duration of monocycle pulse and setting of frame un it time(Τ ) according to multi user numbers and design proper pulse repetition number by transfer rate in multiple access systems design. In the IR system that needs high speed transmission more than 50 Mbps in multiple access interference environment, we could know that very serious performance decrease by multiple access interference happens. Therefore, as the design of high speed multiple access IR system, it should be designed to additional improvement techniques that can remove multiple access interference at the same time.

In this paper, we propose a spatially adaptive image restoration algorithm using local statistics. The local mean, variance, and maximum values are utilized to constrain the solution space, and these parameters are computed at each iteration step using partially restored image. A parameter defined by the user determines the degree of local smoothness imposed on the solution. The resulting iterative algorithm exhibits increased convergence speed when compared to the non-adaptive algorithm. In addition, a smooth solution with a controlled degree of smoothness is obtained. Experimental results demonstrate the capability of the proposed algorithm.

This paper presents a dual-rate (2.4/4.0 kbps) Improved Harmonic-CELP(IHC) speech coder based on the EHC(Efficient Harmonic-CELP) which was presented by the authors. The proposed IHC employs the harmonic coding for voiced and the CELP for unvoiced segments. In the IHC, an initial voiced/unvoiced estimate is obtained by the pitch gain and energy. Then, the final V/UV mode is decided by using the frame energy contour. A new harmonic estimation combining peak picking and delta adjustment provides a more reliable harmonic estimation than that in the EHC. In addition, a noise mixing scheme in conjunction with an improved band voicing measurement provides the naturalness of the synthesized speech. To demonstrate the performance of the proposed IHC coder, the coder has been implemented and compared with the 2.0/4.0 kbps HVXC(Harmonic excitation Vector Coding) standardized by MPEG-4. Results of subjective evaluation showed that the proposed IHC coder and produce better speech quality than the HVXC, with only 40% complexity of the HVXC.

Space-time coding was designed for an efficient transmit diversity technique to improve performance of wireless communication. For the transmit diversity using space-time coding, the receiver requires to estimate channel parameters corresponding to each transmit antennas. In this paper, we propose an efficient channel estimation scheme based on trigonometric polynomial approximation in OFDM systems with transmit diversity using space-time coding. The proposed scheme is more efficient than the conventional scheme in terms of the computational complexity. For QAM modulation, when the size of FFH is 128, the conventional scheme with significant tap caching of 7 requires 9852 complex multiplications for TU, HT and BU channels. But the proposed scheme requires 2560, 7680 and 3584 complex multiplications for TU, HT and BU channels, respectively. Especially, for channels with smaller Doppler frequency and delay spreads, the proposed scheme has the improved BER performance and complexity. In addition, we evaluate the performance of maximum delay spread estimation in unknown channel. The performance of the proposed scheme is investigated by computer simulation in various multi-path fading environments.

Error control and concealment in video communication is becoming increasingly important because transmission errors can cause single or multiple loss of macroblocks in video delivery over unreliable channels such as wireless networks and the internet. This paper describes a temporal error concealment by postprocessing. Lost image blocks are overlapped block motion compensated (OBMC) using median of motion vectors from adjacent blocks at the decoder. The results show a significant improvement over zero motion error concealment and other temporal concealment methods such as Motion Vector Rational Interpolation or Side Match Criterion OBMC by 1.4 to 3.5㏈ gain in PSNR. We present experimental results showing improvements in PSNR and computational complexity.

Application of the Filtered-X LMS adaptive filter to active noise control requires to estimate the transfer characteristics between the output and the error signal of the adaptive canceler. In this paper, we derive an adaptive control algorithm and analyze its convergence behavior when the acoustic noise is assumed to consist of multiple sinusoids. The results of the convergence analysis of the Filtered-X LMS algorithm indicate that the effects of parameter estimation inaccuracy on the convergence behavior of the algorithm are characterize by two distinct components : Phase estimation error and estimated magnitude. In particular, the convergence of the Filtered-X LMS algorithm is shown to be strongly affected by the accuracy of the phase response estimate. Simulation results of the algorithm are presented which support the theoretical convergence analysis.

In this paper, we propose a new MLP based detector which has low circuit complexity and fast adaptation capability for CDMA downlink in frequency selective fading, and is easy for parameter optimization. The simplified structure of the proposed MLP is designed by making use of transmission characteristics of downlinks such that all users signals transmitted over same propagation paths and the number of channelization codes are limited. Significant performance improvement over Rake receiver can be obtained with the proposed MLP and the efficiency of the proposed MLP was compared with that of conventional MLP.

In this paper we propose a blind decision feedback equalizer (DFE) that is characterized by the fact that it does not require channel estimation. Because the output of the optimized multistep prediction error filter (PEF) can be represented as a product of the channel partial impulse response and the transmitted sequence, a backward multistep PEF can be used as the blind DFE feedforward filter (FFF). The corresponding feedback filter (FBF) is obtained from the symbol -rate partial channel impulse response. The proposed algorithm has several advantages over existing blind channel estimation techniques, including stable performance without the necessity of exact channel order estimation.

In this paper, the wavelet image coder, that can encode the image to various bit rate with minimum memory usage, is proposed. The proposed coder is used the 2D significant coefficient array(SCA) that has bit level informal on of the wavelet coefficients to reduce the memory requirement in coding process. The 2D SCA is two dimensional data structure that has bit level information of the wavelet coefficients. The proposed algorithm performs the coding of the significance coefficients and coding of bit level information of wavelet coefficients at a time by using the 2D SCA. Experimental results show a better or similar performance of the proposed method when compared with conventional embedded wavelet coding algorithm. Especially, the proposed algorithm performs stably without image distortion at various b it rates with minimum memory usage by using the 2D SCA.

In this paper, algorithm for non-iterative decoding method is proposed and fractal image decoder based on non-iterative fractal decoding algorithm used general purpose digital signal processors is designed and implemented. The algorithm is showed that the attractor image can be obtained analytically whe n the image is encoded using the fractal algorithm proposed by Monro and Dudbridge, in which the corresponding domain block for a range block is fifed. Using the analytical formulas, we can obtain the attractor image without iteration procedure. And we get general formulas of obtained analytical formulas. Computer simulation results for various test images show that we can increase the image decoding speed by more than five times when we use the analytical formulas compared to the previous iteration methods. The fractal image decoder contains two ADSP2181's and perform image decoding by three stage pipeline structure. The performance tests of the implemented decoder is elapsed 31.2ms/frame decoding speed for QCIF data when all the frames are decoded. The results enable us to process the real-time fractal decoding over 30 frames/sec.

In the general multiple video object coder, more interested objects such as speaker or moving object is consistently coded with higher priority. Since the priority of each object may not be fixed in the whole sequence and be variable on frame basis, it must be adjusted in a frame. In this paper, we analyze the independent rate control algorithm and global algorithm that the QP value is controled by the static parameters, object importance or priority, target PSNR, weighted distortion. The priority among static parameters is analyzed and adjusted into dynamic parameters according to the visual interests or importance obtained by camera interface. Target PSNR and weighted distortion are proportionally derived by using magnitude, motion, and distortion. We apply those parameters for the weighted distortion control and the priority-based control resulting in the efficient bit-rate distribution. As results of this paper, we achieved that fewer bits are allocated for video objects which has less importance and more bits for those which has higher visual importance. The duration of stability in the visual quality is reduced to less than 15 frames of the coded sequence. In the aspect of PSNR, the proposed scheme shows higher quality of more than 2d13 against the conventional schemes. Thus the coding scheme interfaced to human- eye proves an efficient video coder dealing with the multiple number of video objects.

Abstract In this paper, an improved index assignment procedure is proposed to reduce the channel error effect in a communication system employing classified vector quantization(CVQ). The proposed algorithm consists of two parts: inner index assignment (IIA) and cross index assignment (CIA). The II A reduces the distortion resulting from the error in order bits, presenting the identity of each code vector in a subcodebook. The CIA modifies the indexes assigned by the IIA in such a way that the effect of the channel error occurring in class bits, indicating the class information of the code vector, can be minimized. Simulation results show that the proposed algorithms enable a reliable communication over noisy channels even without employing the channel encoding. Index Terms Classified vector quantization, index assignment.

Cellular Automata is discrete dynamical systems which natural phenomena may be specified completely in terms of local relation. In this Paper we Propose noise removal and edge detection algorithm using a Potts Automata which is based on Cellular Automata. The proposed method is aimed to locally increase or decrease the differences in gray level values between pixel of the image without loss of the main characteristics of the image. The dynamical behavior of these automata is determined by Lyapunov operators for sequential and parallel update. We have found that proposed automata rules Present very fast convergence to fixed points, stability in front of random noisy images. Based on the experimental results we discuses the advantage and efficiency.

Noise predictive maximum likelihood(NPML) detector embeds noise prediction/whitening process in branch metric calculation of Viterbi detector and improves the reliability of branch metric computation. Therefore, PRML detector with a noise predictor achieves some performance improvement and has an advantage of low complexity. This thesis random sequences are applied to linear channel. In perpendicular magnetic recording density KP=2.5, NP(121)ML and NP(1221)ML detection system which is based on a noise predictive PR-equalized signal are evaluated by the Performance through a computing simulation. Therefore, NPML systems are implemented and are verified by VHDL.